We saw how to develop a WebRTC client, install an SIP server, and configure a WebRTC to SIP environment. A sky view of our final, existing client-server solution setup for SIPWS signaling and WebRTC media so far is shown as follows:
As per our current setup status, only the WebRTC-enabled client and servers can participate in the communication flow in an offer/answer (O/A) model.
There are, however, numerous limitations of the existing solution, some of which are mentioned in the following sections. In the upcoming chapters, we shall do away with most of the limitations.
The existing architecture does not provide the Network Address Translation (NAT) technique to overcome the blockage due to firewalls and enterprise policies. As a solution, we must see the alternative for public IP discovery in the WebRTC client server setup. NAT is possible in the Kamailio server through RTP proxy modules and STUN.